1. Field of the Invention
The present invention relates generally to a mobile communication system supporting a voice service via a packet network. More particularly, the present invention relates to a method and apparatus for controlling a rate of a particular voice service.
2. Description of the Related Art
Mobile communication systems are evolving into high-speed, high-quality wireless data packet communication systems for providing not only the established voice-oriented service, but also data and multimedia services. A Universal Mobile Telecommunication Service (UMTS) system, which is the 3rd generation mobile communication system that is based on Global System for Mobile Communications (GSM) and General Packet Radio Services (GPRS) and uses Wideband Code Division Multiple Access (CDMA), provides a service in which users of mobile phones or computers can transmit packet-based text, digitized audio/video and multimedia data at a high rate of 2 Mbps or higher anywhere in the world. The UMTS system employs a packet-switched access concept that uses a packet protocol such as Internet Protocol (IP).
In this context, in the 3rd Generation Partnership Project (3GPP) that is responsible for standardization for the UMTS communication system, there is a discussion on Voice over Internet Protocol (VoIP) communications that support voice packets using Internet protocol in supporting a voice service.
VoIP refers to a communication technique for transmitting voice frames generated in a voice codec after converting the voice frames into IP/User Datagram Protocol (UDP)/Real time Transport Protocol (RTP) packets. With the use of VoIP, it is possible to provide voice service via the packet network.
FIG. 1 is a diagram illustrating a configuration of a conventional mobile communication system supporting VoIP.
Referring to FIG. 1, a user equipment (UE) 100 includes a codec 105 for converting a voice signal into a voice frame, an IP/UDP/RTP protocol layer 104 for making an IP/UDP/RTP packet with the voice frame from the codec 105, a Packet Data Convergence Protocol (PDCP) layer 103 for compressing a header of the IP/UDP/RTP packet, a Radio Link Control (RLC) layer 102 for converting the IP/UDP/RTP packet into an appropriate format to transmit it over a wireless channel, and a Medium Access Control (MAC)/Physical Layer (PHY) layer 101 for transmitting the packet data over the wireless channel.
The voice packet data transmitted from the UE 100 is delivered to a radio network controller (RNC) 120 via a Node B 110 over a wireless channel. The RNC 120, including a MAC/PHY layer 121, an RLC layer 122 and a PDCP layer 123, like the UE 100, converts the voice frame on the wireless channel into its original IP/UDP/RTP packet, and transmits the IP/UDP/RTP packet to a core network (CN) 130. The IP/UDP/RTP packet delivered to the CN 130 is transmitted to a receiving UE (not shown) via an IP network 140. The receiving UE, having the same hierarchical structure as that of the transmitting UE 100, restores the transmitted voice data to its original voice signal in the opposite order.
3GPP provides voice service using a voice codec called an Adaptive Multi Rate (AMR) codec, and the AMR codec is characterized by its variable rate, so it can adjust the rate according to conditions of the wireless channel.
FIG. 2 is a diagram illustrating a conventional AMR rate control process.
In a downlink (DL) rate control process of FIG. 2, both ends of a VoIP communication exchange VoIP packets 265 with each other. The VoIP packet 265 is composed of an IP/UDP/RTP header 240, an AMR payload specific header 257 and voice data 260. The AMR payload specific header 257 is composed of a Change Mode Request (CMR) field 245, Frame Type (FT) field 250 and an OTHER field 255.
The voice data 260 can have various sizes according to an AMR codec mode, and the FT field 250 is filled with information on the codec mode applied to the voice data. Therefore, a VoIP packet receiving party can ascertain a size and format of the voice data 260 by parsing the information in the FT field 250. Table 1 below shows exemplary sizes of voice data according to various codec modes used in the AMR.
TABLE 1Frame TypeAMR codec modeVoice data size (bits)0AMR 4.75 kbit/s1121AMR 5.15 kbit/s1202AMR 5.90 kbit/s1283AMR 6.70 kbit/s (PDC-EFR)1444AMR 7.40 kbit/s (TDMA-EFR)1605AMR 7.95 kbit/s1766AMR 10.2 kbit/s2167AMR 12.2 kbit/s (GMS-EFR)2568AMR SID56
The CMR field 245 is filled with information on a codec mode required for the other party. For example, if the other party desires to transmit voice data at 12.2 kbps, the CMR field 245 is filled with 7, which is an FT value associated with 12.2 kbps.
The OTHER field 255 is filled with such information indicating whether there is an error in the voice data. This field is not related to the gist of the exemplary embodiment present invention, so a description thereof will be omitted.
A description will now be made of a downlink rate control process performed in a UMTS network.
A VoIP communication is being performed between a UMTS terminal 205 and an IP entity 210 via an RNC 207. The IP entity 210 is a device for generating downlink VoIP packets, and can include, for example, a gateway or a VoIP terminal. In step 215, the IP entity 210 transmits a VoIP packet having a rate of y kbps to the UMTS terminal 205. In this case, an FT field of the VoIP packet is filled with a value x associated with the rate. In step 220, the UMTS terminal 205 desires to change the rate of the downlink VoIP packet. Exemplary reasons for the UMTS terminal 205 to change the rate of the downlink VoIP packet include, if there is a need to decrease the rate due to the deterioration of a downlink wireless channel, or to increase the rate due to the improvement of the downlink wireless channel.
In step 225, the UMTS terminal 205 inserts an FT value associated with a desired rate into a CMR field of an uplink (UL) VoIP packet before transmission. In step 230, upon receipt of the uplink VoIP packet, the IP entity 210 changes a rate of a downlink VoIP packet to x−1 according to a CMR field value. Thereafter, an FT of the downlink VoIP packet transmitted to the UMTS terminal 205 in step 235 becomes x−1.
Because the rate control in the conventional VoIP communication is performed on an end-to-end basis, the entity located in the middle of the transmission path cannot control the rate. However, in the UMTS network, there is a case where an RNC should be able to control an AMR rate so as to be closely associated with radio resources. For example, there is a possible case where the RNC should decrease a downlink/uplink AMR rate in order to limit to a specific level a ratio of VoIP traffic to all traffic of a particular cell. However, in conventional VoIP communications, the RNC cannot control the AMR rate.
Accordingly, there is a need for an improved method and apparatus for controlling a rate of a voice service in a mobile communication system supporting the voice service via a packet network